similar to: Preventing an extension from dialing certain outbound codes

Displaying 20 results from an estimated 200 matches similar to: "Preventing an extension from dialing certain outbound codes"

2005 Sep 14
1
Indications for Ireland
Hello asterisk-users, Just curious if anyone has the indications for Ireland, tried googling for it to no avail. Sean -- +---------------------------------------------------+ |VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie | |GPG Key http://thecivvie.fastmail.fm/thecivvie.asc | +---------------------------------------------------+ Strange things happen under the midnight sun when Men
2005 Aug 19
1
Console OSS or ALSA for 3.4
What do I need to install for OSS or ALSA for console use only Sean -- +----------------------------------------------------+ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | +----------------------------------------------------+ -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type:
2005 Aug 14
2
Problem with FWD connection rejected
Using the install instructions for A@H, I setup a FWD account, this I tested using X-Lite and it works okay, Nowever I cannot make calls to fwd using Asterisk, my log showes: Aug 14 21:06:09 NOTICE[1324]: Registration of '689482' rejected: Registration Refused Aug 14 21:06:59 NOTICE[1324]: Registration of '689482' rejected: Registration Refused According the to info screen of
2005 Aug 15
7
8 FXS in Asterisk Server
Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones to. The problem is, that there is only one PCI slot available. What i have is 4 USBs 2.0 interfaces free (if this helps). So here's my question: how am i going to do this? i tried to find any PCI cards supporting 8 FXS interfaces, but without success. does anyone know such hardware? Thanks in
2005 Aug 11
8
Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone
I have a brief from a local hotel to build a PBX using Asterisk but they want to use their exisiting telephones and wiring from an old PBX that no longer works. Basically, I can build the system but an looking for a card that will allow for upto 20 extensions to be wired into the back of the PC. Doeas anyone know of a solution to this Sean-- ICQ: 679813 FidoNet: 2:263/950 Jabber:
2005 Sep 20
1
[3.5 Server] Problem with libjpeg.a or absence of
Hello CentOS, I am trying to compile zoneminder which requires libjpeg.a I jave libjpeg.so installed okay. Is there a package or do I have to recompile the libjpeg package Sean -- +---------------------------------------------------+ |VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie | |GPG Key http://thecivvie.fastmail.fm/thecivvie.asc |
2005 Aug 13
3
One more newbie question
Ok, I am going for A@H with the CentOS iso disk. Installed and just checking a few things out. My other question is this, which I forgot to ask before. We have no Broadband here and more than likely will never have, so I am just looking at building Asterisk to handle inbound and outbound calls, at home via a ISDN card and for my hotel job via a PSTN line setup. Is this very complicated to setup
2003 Mar 07
70
unsubscribe
Gautham Kasinath Software Engineer Arkin Systems Pvt Ltd T. Nagar Chennai Ph. (91) (44) 8216686 Extn 14
2008 Mar 27
6
Problems pinging PC on tunnel
Hello! I have set up tunnel between a FreeBSD machine and Windows Vista. Tunnel is established, but when I try to ping either end ping fails. I have temporarily switched off firewalls on both machines, no luck. Here is client tinc.conf on Vista: Name = lenovo_client ConnectTo = lenovo_server Interface = tinctap Subnet = 10.20.40.0/24 Sevrer tinc.conf on FreeBSD: Device=/dev/tap0
2006 Nov 10
2
A new attack
Log report is reporting a lot of these lately.. following is just a short snippet from the beginning on one server. WARNING!!!! Possible Attack: Attempt from 104.29.broadband2.iol.cz [83.208.29.104] with: command=HELO/EHLO, count=3 : 1 Time(s) Attempt from 106.7.broadband7.iol.cz [88.102.7.106] with: command=HELO/EHLO, count=3 : 1 Time(s) Attempt from
2005 Aug 11
2
Newbie Question: Building an Asterisk systemto replace an old PBX but using existing phone
Yeah....I think that every install I have done the first thing that happens is "why is there a delay before the call connects?" and the answer is "you have to hit dial or wait 10 seconds". -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Rymes Sent: Thursday, August 11, 2005
2012 Sep 20
3
Sendmail log entries
Recently we began seeing lots of these log entries on our off-site mx smtp host. I have googled this but I am not clear from what I have read if this is something we can stop altogether or should even worry about. Comments? Logwatch. . . --------------------- sendmail Begin ------------------------ SMTP SESSION, MESSAGE, OR RECIPIENT ERRORS ------------------------------------------
2002 Oct 12
1
replacing Courier imapd
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I currently use courier imapd, exim and ldap on my mail box. There are no users on this box, all info is taken from the ldap server. I note that dovecot does not use ldap as a backend, but curious if this in on the cards, or even mysql backend. Sean - -- Sean Rima http://www.tcob1.net Linux User:
2005 Sep 10
2
VoipBuster again
Hi, all I am still battling to connect * and voipbuster. What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or IAX traffic when using their client. VoipBuster client connects to connectionserver.voipbuster.com on port 11112 for authentication. Call itself is placed on different server. I have tried to connect using SIP and IAX and it seems that no authentication is
2006 May 29
4
registration at Voipbuster times out
Hi, I am new here on this list, and have a problem of which I hope that somebody here can help me with it. I have a Voipbuster account, with which I would like to make phone calls via my Asterisk PBX. If I let X-Lite register directly at voipbuster.com, everything is OK, but if I let Asterisk register there, it says "registration for XXXXXX@sip.voipbuster.com timed out, trying again",
2005 Aug 31
7
VoipBuster with astersisk?
Hi, all Here is a something I found on the web: http://www.voipbuster.com And it works OK too. Now, I want to use it via asterisk, so I ccan use my normal phones instead of PC application. Did anyone try to connect astersisk and VoipBuster? Thanks, Rudolf
2009 Sep 02
1
Voipbuster not ringing, other SIP peers are ringing...
Does anybody else see the same behavior for VoipBuster connections? When I trace one of the other SIP peers, I see it sends this message: ---------------------------------------------------------------------- <--- SIP read from 82.101.62.99:5060 ---> SIP/2.0 180 Ringing Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE Call-ID: 740540ee64fa957513ce89f03ef5e3f2 at sip.xs4all.nl
2005 Sep 26
1
voipbuster advise
Hi, I'm using voipbuster at work, and I've got 2 questions: 1) Is it possible to send faxes using voipbuster connex? 2) Is it possible to cut off or cover the voice that say the charge per minute?(I've payed the '5' euro, and from that moment I've got it!). Of course I understand that is to let me know how much I'm going to spend, but I do not like it, expecially when
2006 Jan 19
0
Incoming fax on voipbuster
Hello, I'm trying to receive a fax to my inbound number from voipbuster. Asterisk receives the call and starts the rxfax application successful, but then nothing happens. The calling party is still hearing a ringing tone, or sometimes nothing. Voicecalls are working correct and without problems. For testing I've add a local number (300) to the dialplan. When I call this number
2006 Jun 08
0
"I can hear them but they can't hear me" with VoipBuster
Hi;? When connecting via VoipBuster or VoipStunt, "I can hear them but they can't hear me"?. This happens with VoipBuster or Voipstunt. Registration is done correctly. I thought it could be something related to NAT, but I don't have this problem when using VoipUser or Asterisk2PSTN, for example. ? I tried with different codecs: gsm, alaw and ulaw but no change. ? So, now?I